Low delay corrector

ABSTRACT

A low delay corrector (LDC) unit includes a non-linear function generator and a filter. The nonlinear function generator receives a first signal and outputs a second signal in dependence on the first signal and a transfer function of the nonlinear function generator. The filter is fed in dependence on the second signal output by the nonlinear function generator. The first signal received by the nonlinear function generator is derived in dependence on an input signal provided to an input of the LDC unit and an output of the filter. An output of the LDC unit is derived in dependence on the first signal received by the nonlinear function generator and the second signal output by the nonlinear function generator.

PRIORITY CLAIM

This application is a Divisional of U.S. patent application Ser. No.11/782,708, by Chieng et al., entitled “Digital PWM Amplifier Having aLow Delay Corrector”, filed Jul. 25, 2007, which claims the benefit ofU.S. patent application Ser. No. 11/324,132, by Andersen et al.,entitled “Digital PWM Amplifier with Simulation-based Feedback,” filedDec. 30, 2005 (now U.S. Pat. No. 7,286,009), each of which isincorporated by reference as if set forth herein in its entirety.

RELATED APPLICATION

This application relates to U.S. patent application Ser. No. 11/782,702,by Andersen et al., entitled “Low-Noise, Low-Distortion Digital PWMAmplifier”, filed Jul. 25, 2007, which is incorporated by reference asif set forth herein in its entirety.

FIELD OF THE INVENTION

The invention relates generally to a low delay corrector that can beused in digital amplifiers, such as a digital switching power amplifier,as well as in other systems.

BACKGROUND

Practical audio power amplifiers using Pulse Width Modulation (PWM) havebeen known since the mid 1960s. In amplifiers from that era, a pulsetrain was generated by comparing a voltage representing the incomingaudio signal with a reference waveform, typically a triangular wave orsawtooth wave, with a frequency in the range 50 kHz-200 kHz. Thecomparison yielded a 2-level rectangular wave having the same frequencyas the reference waveform, and with a mark:space ratio varying insympathy with the audio. The rectangular wave was amplified to thedesired power level and then passively lowpass filtered to remove mostof the high-frequency components of the rectangular wave, leaving itsaverage level, which follows the audio, to drive a load such as aloudspeaker.

It is possible to obtain extremely good performance when such amplifiersare run ‘open-loop’, that is without feedback, but it is an expensivesolution since the amplifier's performance is critically dependent onthe quality of the output stages and the power supply. To alleviatethese dependencies, the trend in the 1970s and subsequently has been toincorporate feedback. One simple way to incorporate feedback in anamplifier that compares the audio with a triangle wave, is to replace afixed triangle wave by a sawtooth wave that is obtained by integratingthe substantially rectangular wave that appears at the output of theamplifier's power switches. Analysis shows that this is an effectivemeans of providing feedback. Moreover since the feedback is tightlyintegrated into the PWM itself, stability problems typically associatedwith feedback do not arise.

Amplifiers as described above have sometimes been called ‘digital’ inthe popular press, but we shall describe them as ‘analog’, because thetimings of the edges of the rectangular waves can vary continuously insympathy with the audio. We shall reserve the word ‘digital’ for anamplifier in which the edge timings are quantized, so that the edgetimings can be represented digitally and the edges can be generated bycounting pulses produced by a high-precision, high-frequency clock, suchas a crystal oscillator.

This principle was proposed by Sandler [6], who also realized that theapparent need for a clock frequency in the gigahertz region could beavoided by the use of oversampling and noise shaping. Several commercialproducts are now available that use this principle (see, for example,[3].)

The digital principle brings precision to the generation of the PWMwaveform, but the power amplification, typically accomplished by MOSFET(Metal Oxide Silicon Field Effect Transistors) power switches, remains afundamentally analog process, and as such is vulnerable to non-idealcomponent behavior. There is a distortion associated with the switchingcalled “dead-time distortion”, and there is dependency on the powersupply just as with the original analog PWM amplifiers. Without feedbackor other compensation, the gain of the output stage will be directlyproportional to the supply voltage. This precludes the use of aninexpensive non-regulated power supply, or condemns the system torelatively poor performance.

Attempts have been made in the prior art to apply feedback to the outputstages of a digital PWM amplifier. One such attempt is embodied in thePEDEC (PCT/DK98/00133) principle, in which a modulator operating at arelatively low level produces a PWM waveform, and a correction unitre-times the edges of the waveform before passing the waveform to thepower switches. The correction unit receives control signals from anerror processing unit, which compares the original low-level PWMwaveform with the output of the power switches. The input to the powerswitches is thus modified in dependence on the output, creating afeedback loop.

The PEDEC principle can be applied to a digital or an analog PWMamplifier. However the feedback is analog and local to the outputstages—the quality of the output is fundamentally determined by thequality (including jitter properties) of the low-level PWM waveform.

Another example of feedback in the prior art is the disclosure byMelanson in U.S. Pat. No. 6,373,334 “Real Time Correction of a DigitalPWM Amplifier”. Again, the feedback is derived by comparing a low-levelsquare wave with the output of the power switches. In this proposal,however, the correction is fed back to the PWM modulator, so there donot exist two PWM waveforms, original and re-timed, as in the PEDECproposal. U.S. Pat. No. 6,373,334 describes a feedback that is tightlyintegrated into a particular type of PWM modulator. It shares with PEDECthe property that the quality of the final output is limited by thequality of the low level PWM waveform.

In an analog (non-PWM) amplifier, it is customary to take at least somefeedback from the final output to a point close to the input. Asubstantial reason why this is difficult in a digital PWM amplifier isloop delay. In particular, since the output is analog but the input andearly processing are digital, an ADC (Analog to Digital Converter) isrequired in the feedback path. Depending on the topology, the quality ofthe final output will be directly related to the quality of the ADC.Currently available audio ADCs of sufficient quality, however, havedelays that are completely excessive for inclusion in a loop thatprovides significant feedback over the audio range of 0-20 kHz.

Even when the ADC delay has been minimized, substantial stabilityproblems remain. There is an extensive literature on stabilizingfeedback loops, using Bode plots, lead/lag compensation, nested feedbackand the like. Most of the techniques apply to linear systems withconstant gain, and there is little guidance on how to deal withnonlinearity or gain variation apart from allowing an adequate “gainmargin” or “phase margin”.

Unfortunately, a loop that includes a delay of, for example, 10 μs, andthat has enough “gain margin” or “phase margin” to be robust againstnonlinearity and gain variation, is unlikely to provide a significantdegree of feedback at 20 kHz. “Nested feedback” appears at first sightto be able to provide large amounts of feedback with stability. Onexamination, however, it is found that the stability is “conditional”,which means that it is susceptible to gain variation, and oscillationcan be caused even by a reduction of the gain of the forward path.Consequently, this technique would be completely unsuitable for use in aPWM amplifier that is required to work with an unregulated power supply.

A less obvious problem is the intrinsic nonlinearity introduced by thepulse width modulation process. This is normally thought of as a smalleffect that introduces harmonic distortion at high audio frequencies(e.g., −70 dB 3rd harmonic on a full scale 5 kHz fundamental [3].)However, design of a feedback loop requires one to consider frequencieswell outside the band that is effectively controlled by feedback. In thecase of a digital PWM amplifier with a sampling and switching frequencyof 384 kHz, frequencies up to the Nyquist of 192 kHz should ideally beconsidered. At 192 kHz, the forward gain of a conventional double-edgePWM modulates by 100% as the mark:space ratio of the PWM waveform variesover its full range. Even at 80 kHz, the forward gain modulates by 20%.Such modulation of a part of the spectrum that is only two octaves abovethe top of the range that is desired to be controlled will set a limitto how “aggressive” any conditionally stable feedback can be, even foramplifiers that are always used with stabilized power supplies.

Several correction methods are known for PWM nonlinearity. Onestraightforward method, as shown in [3], achieves almost completecancellation of the nonlinear effect within the audio band. However ifit is hoped that feedback stability will be improved by correcting thePWM nonlinearity, then the corrector must be placed inside the feedbackloop. Since the corrector in [3] has a delay of one sample (e.g. 2.6 μs)the stability problem is already worse. Further, while the correction isalmost perfect within the audio band, it still does not provideconsistent performance near the Nyquist frequency, for it is notpossible to compensate a gain modulation of 100%.

In view of the difficulties discussed above, there is a need for arobust method for applying feedback to a digital PWM amplifier thatdirectly addresses the issues of loop delay, nonlinearity and variationin the forward gain.

SUMMARY

This disclosure is directed to systems and methods for performanceimprovements in a digital switching power amplifier using a low delaycorrector. In the various embodiments of the present invention, theoutput of an output stage is sampled, converted to a digital signal, andfed to a low delay corrector, with the output of the low delay correctorbeing fed back into the audio signal (e.g., prior to a noise shaper.)The low delay corrector is configured to substantially correct a portionof the nonlinear effects of the pulse width modulator over an operatingfrequency range.

In an exemplary embodiment, a digital pulse width modulation (PWM)amplifier includes a signal processing plant configured to receive andprocess an input audio signal. The amplifier also includes a low delaycorrector configured to receive signals output by the plant. The outputof the low delay corrector is added to the input audio signal asfeedback. In various embodiments, the plant may consist of a modulatorand power switch, a noise shaper, or any other type of plant. If theinput of the plant is digital and the output is analog, ananalog-to-digital converter (ADC) is provided to convert the outputaudio signal to a digital signal. Low-pass filtering may be implementedbefore or after the ADC, and a decimator may be placed after the ADC ifit is an oversampling ADC. A simulator may perform linear or nonlinearprocessing on the audio signal or may introduce delays into the signalas needed to simulate the plant.

In one embodiment, a switching amplifier employing a digital pulse widthmodulator and power switches that feed an output is provided with asimulator that models the behavior of the modulator and/or of the powerswitches, and with a subtractor that derives an error signal independence on the difference between the output of the simulator and theoutput of the power switches. The input to the pulse width modulator ismodified by a feedback signal in dependence on the difference, as wellas the output of the low delay corrector.

In one embodiment, the signal is filtered by a substantiallyminimum-phase filter whose response rises above an operating frequencyrange, in order to provide phase advance that compensates some of theassociated delay.

In one embodiment, the amplifier contains calibration and adjustmentunits that act to minimize the difference signal. Preferably, gaindifferences between the two inputs to the subtractor will becompensated, and typically this is done by adjusting the gain of thefeedback path or the simulator. In some embodiments, delay differencesbetween the two paths will also be monitored and compensated. Typically,the calibration unit receives the difference signal, detects anycorrelation between the error signal and the input to the feedback loop,and requests an adjustment that will reduce that correlation.

In one embodiment, the amplifier includes, prior to the main feedbackloop, a predistortion unit that substantially compensates the nonlineareffects of the pulse width modulator that have not been compensated bythe low-delay corrector. In some embodiments, the input to thepredistortion unit is modified by low frequency components of thefeedback signal.

In another embodiment, a switching amplifier is provided with a feedbackpath that includes an ADC whose input is responsive to the differencebetween a signal derived from a low-level PWM waveform and a signalderived from the output of power switches. Typically, the ADC is of anoversampling type, is preceded by an analog lowpass filter and isfollowed by a decimator. Typically, the feedback path includes a digitalshaping filter whose response rises above the operating frequency rangein order to compensate, within the operating frequency range, delays inthe feedback loop. Typically, the feedback loop includes also alow-delay corrector that provides approximate or substantial correction,over the operating frequency range, for the nonlinear behavior of apulse width modulator.

In another embodiment, a switching amplifier is provided with a feedbackpath comprising an oversampling ADC followed by a decimation filter anddecimator producing a decimated output. The decimation filter issubstantially minimum phase and has an amplitude response that istailored to provide, at each frequency above the Nyquist frequency ofthe decimated output, substantially the minimum attenuation required inorder to reduce the aliased image of that frequency to an acceptablelevel. Typically, the decimator filter is an FIR filter some of whosezeroes are not configured to provide maximum attenuation at the samplingfrequency of the decimated output or its harmonics.

Numerous other embodiments are also possible.

BRIEF DESCRIPTION OF THE DRAWINGS

Other objects and advantages of the invention may become apparent uponreading the following detailed description and upon reference to theaccompanying drawings.

FIG. 1 is a digital pulse width modulated amplifier in accordance withthe prior art.

FIG. 2 is a diagram illustrating small signal amplitude response ofdouble edge (class AD or class BD) PWM modulator operating at 384 kHz,with pulse width, as a percentage of the pulse repetition period, asparameter.

FIG. 3 is a diagram illustrating a digital pulse width modulatedamplifier including feedback according to one embodiment of theinvention.

FIG. 4 is a diagram illustrating the internal structure of anoversampling ADC in one embodiment.

FIG. 5 is a table showing coefficients tap[0] through tap[79] of an80-tap FIR decimation filter in one embodiment.

FIG. 6 is a diagram illustrating the amplitude response of an 80-tap FIRdecimation filter in one embodiment.

FIG. 7 is a diagram illustrating the amplitude response of an 80-tap FIRdecimation filter and (dashed) cascade of four combs in one embodiment.

FIG. 8 is a diagram illustrating the alias attenuation of an 80-tap FIRdecimation filter and (dashed) cascade of four combs in one embodiment.

FIG. 9 is a diagram illustrating z-plane zeroes of a comb filter witheight equal taps in one embodiment.

FIG. 10 is a diagram illustrating z-plane zeroes of an 80-tap FIRdecimation filter in one embodiment.

FIG. 11 is a diagram illustrating a close-up of five z-plane zeroes ofan 80-tap FIR decimation filter near to z=0+1i in one embodiment.

FIG. 12 is a diagram illustrating a measurement path in one embodiment.

FIG. 13 is a diagram illustrating a low-frequency model of PWMnonlinearity in one embodiment.

FIG. 14 is a diagram illustrating the response of an analog lowpassfilter to a PWM pulse in one embodiment.

FIG. 15 is a diagram illustrating the sharpening of an analog filterresponse using a 3-point deconvolution filter in one embodiment.

FIG. 16 is a diagram illustrating the response of an 80-tap FIRdecimation filter to sharpen analog filter response in one embodiment.

FIG. 17 is a diagram illustrating a conceptual model of a measurementpath in one embodiment.

FIG. 18 is a diagram illustrating a practical simulator architecture inone embodiment.

FIG. 19 is a diagram illustrating detail of an amplifier incorporatingan alternative simulator in one embodiment.

FIG. 20 is a diagram illustrating the amplitude response of a predictionfilter H′ in one embodiment.

FIG. 21 is a table showing coefficients H[0] through H[24] of feedbackfilter H, implemented as 25-tap FIR in one embodiment.

FIG. 22 is a diagram illustrating the amplitude response of the feedbackfilter H of FIG. 21.

FIG. 23 is a diagram illustrating the magnitude of gain of feedback loopwith the simulator disabled in accordance with one embodiment.

FIG. 24 is a diagram illustrating the magnitude of the noise transferfunction (NTF) of a feedback loop in one embodiment.

FIG. 25 is a diagram illustrating a low delay corrector unit LDC in oneembodiment.

FIG. 26 is a diagram illustrating the small-signal amplitude response ofthe low delay corrector unit of FIG. 25, with pulse width as aparameter.

FIG. 27 is a diagram illustrating a Gerzon first-order predistortionapplied to nonlinear element N that approximates delay in oneembodiment.

FIG. 28 is a diagram illustrating a Gerzon predistortion unit adapted tocompensate LDC and S in one embodiment.

FIG. 29 is a diagram illustrating a Gerzon predistortion unitincorporating correction for small-signal transfer function in oneembodiment.

FIG. 30 is a diagram illustrating a Gerzon predistortion unit adapted tocompensate LDC and pulse width modulator in one embodiment.

FIG. 31 is a diagram illustrating a Gerzon predistortion unit adapted tocompensate a pulse width modulator, followed by a compensator for LDC inone embodiment.

FIG. 32 is a diagram illustrating the detail of a predistortion unitusing the principle of FIG. 31.

FIG. 33 is a diagram illustrating the detail of a predistortion unitusing the principle of FIG. 30.

FIG. 34 is a diagram illustrating an amplifier with compensation for theeffect of power supply variations in one embodiment.

FIG. 35 is a diagram illustrating an amplifier with reference path inone embodiment.

FIG. 36 is a diagram illustrating joint clipping of the main andreference paths in one embodiment.

While the invention is subject to various modifications and alternativeforms, specific embodiments thereof are shown by way of example in thedrawings and the accompanying detailed description. It should beunderstood, however, that the drawings and detailed description are notintended to limit the invention to the particular embodiment which isdescribed. This disclosure is instead intended to cover allmodifications, equivalents and alternatives falling within the scope ofthe present invention as defined by the appended claims.

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS

One or more embodiments of the invention are described below. It shouldbe noted that these and any other embodiments described below areexemplary and are intended to be illustrative of the invention ratherthan limiting.

As described herein, various embodiments of the invention comprisesystems and methods for systems and methods for performance improvementsin a digital switching power amplifier using a low delay corrector.

In an exemplary embodiment, a digital pulse width modulation (PWM)amplifier includes a signal processing plant configured to receive andprocess an input audio signal. The amplifier also includes a low delaycorrector configured to receive signals output by the plant. The outputof the low delay corrector is added to the input audio signal asfeedback. In various embodiments, the plant may consist of a modulatorand power switch, a noise shaper, or any other type of plant. If theinput of the plant is digital and the output is analog, ananalog-to-digital converter (ADC) is provided to convert the outputaudio signal to a digital signal. Low-pass filtering may be implementedbefore or after the ADC, and a decimator may be placed after the ADC ifit is an oversampling ADC. A simulator may perform linear or nonlinearprocessing on the audio signal or may introduce delays into the signalas needed to simulate the plant.

FIG. 1 shows a typical prior art digital power amplifier employing PulseWidth Modulation, as described more fully in [3]. In FIG. 1, apulse-width modulator furnishes a square wave of variable mark:spaceratio, alternatively described as a sequence of pulses of variablewidth. In the case illustrated, known as “symmetrical class ADmodulation”, the rising and falling edges of the pulse are moved inopposite directions in response to the input to the modulator, asindicated by the arrows in FIG. 1. The modulated pulse sequence is fedto a driver stage (not shown) and then to power switches, typicallyMOSFETS. In a typical implementation, two MOSFETS will be driven inantiphase so that their junction is connected alternately to the powerrails +Vcc and −Vcc and thereby carries a high level PWM waveform whosemark:space ratio is modulated by the signal.

An LC filter is provided both for efficiency reasons and to remove thesquare wave from the final output. The filtered analog output thenfollows, approximately, the input to the modulator, and can be used todrive a load such as a loudspeaker.

Several variants of the topology of FIG. 1 are known in the prior art,including use of four switches in a full “H-bridge” rather than two in ahalf-bridge as shown.

The input to the modulator is digital. In “symmetrical class AD”modulation, each digital sample controls the timings of both edges of apulse. There are also “leading edge” and “trailing edge” modulationschemes in which just one pulse edge is modulated, and also “consecutiveedge modulation” in which, for example, even-numbered input samplescontrol the rising edges and odd-numbered samples control the fallingedges of the pulses. Thus, with consecutive edge modulation (and alsowith “class BD modulation”, also known as “three-level modulation”) thedigital sampling frequency is twice the power switching frequency,whereas with the other modulation schemes mentioned above, the twofrequencies are the same.

Alternative modulation schemes and power switch topologies will not bediscussed further, but it is to be understood that the invention is notrestricted to symmetrical class AD modulation, nor is it restricted tothe half-bridge power switch topology.

A typical sampling frequency for the input to the modulator is 384 kHz.An amplifier receiving a digital input a standard consumer samplingfrequency between 44.1 kHz and 192 kHz will therefore require anupsampler, not shown in FIG. 1.

A digital pulse width modulator determines the timings of the edgetransitions of the PWM waveform by counting the beats of a digitaloscillator or clock. The maximum clock frequency that is practical withcurrent technology is of order 300 MHz, which implies that there arefewer than 1000 distinct pulse lengths possible, or fewer than 500 ifsymmetrical class AD modulation is used. Used directly, this givesdigital resolution lower than that of 9-bit PCM, or a noise floor worsethan −66 dB over the conventional audio frequency range 0-20 kHz, or −56dB as seen over the Nyquist range 0-192 kHz. The purpose of the noiseshaper in FIG. 1 is to requantize the incoming digital audio signalhaving a wordwidth typically between 16 and 28 bits, down to a wordwidthof typically 9 bits or fewer, with a noise level somewhere between −100dB and −135 dB over 0-20 kHz. However, the noise shaping increases thenoise as seen wideband 0-192 kHz by typically up to 12 dB, and it is tobe noted that wideband noise at −44 dB reduces significantly theheadroom for signal excursion before clipping.

Digital pulse width modulation is inherently nonlinear. The nonlinearityhas a precisely known mathematical form and can be corrected to highaccuracy, within the audio range, by a predistortion unit as shown inFIG. 1 and discussed in 3. Such correction by predistortion can beextremely effective, but there remain other problems that cannot becharacterized so accurately in advance, and therefore are not asamenable to correction by predistortion. These problems includedistortion effects such as “dead time distortion” associated with thepower switches, and modulation of the audio by the power supply. It willbe appreciated that the modulator merely varies the proportion of timethat the junction between the two switches spends at +V_(CC) and−V_(CC), so the amplitude of the filtered output waveform will beproportional to the difference (+V_(CC))-(−V_(CC)).

In a typical analog PWM amplifier, these problems are substantiallyreduced by overall feedback, but substantial problems confront theperson who would think to place most or all of the elements shown inFIG. 1 within a feedback loop. Firstly, the power switch output isanalog while the signal at the input to the modulator and at earlierpoints in the chain is digital, so an ADC (analog-to-digital converter)will be needed. It is desired to keep delay to a few microseconds so aspecial type of ADC is needed.

Even with a specially designed fast ADC, it is still difficult toachieve a loop delay lower than 5 μs-10 μs. A delay of 10 μs correspondsto a phase shift of 72° at 20 kHz, and to obtain substantial feedbackover 0-20 kHz using prior art methods, one would be forced to consider a“conditionally stable” design. In such a design, the phase is allowed toexceed 180° at some frequencies at which the (modulus of) loop gain isgreater than unity, but the phase must be brought down to less than 180°when the loop gain makes the transition from being greater than unity tobeing less than unity.

If overload considerations can be handled, a conditionally stablefeedback loop can be satisfactory in the context of a linear, or almostlinear, system. However, the pulse width modulation process isnonlinear. In the case of double-edge modulation the transfer functionis flat in the limit as the pulse length tends to zero, but shows anincreasing high frequency droop as the pulse length increases, as shownin FIG. 2. The amplitude response at the Nyquist frequency (192 kHz inthe case illustrated) tends to zero as the pulse length tends to 100% ofthe pulse repetition period. This implies that a conditionally stablefeedback loop that is stable at zero pulse length is likely to havestability problems as pulse length increases.

In the PEDEC and Melanson prior art already cited, these difficultiesare avoided by applying the feedback to the power switches only, and inthe case of PEDEC by keeping the feedback in the analog domain so thatADC delay is avoided completely.

Embodiments of the present invention address the questions of how tominimize the delay introduced by a high-quality audio ADC, and how tokeep a digital feedback loop stable despite the remaining delay and thePWM nonlinearity.

Amplifier Topology

An exemplary embodiment of the invention will now be described withreference to FIG. 3. Signals to the left of the dashed line in FIG. 3are digital, while those to the right are analog. The PWM modulator andthe ADC are the interface between the two domains. The noise shaper,pulse width modulator, power switches and LC filter in FIG. 3 performthe same function as the corresponding items in FIG. 1.

In order to provide feedback in FIG. 3, the power switch output isfiltered by an analog low pass filter and converted to digital form bythe ADC. The unit LPF⁻¹ provides partial correction for the effect ofthe analog lowpass filter. A subtraction node then subtracts from thefeedback a comparison signal calculated by the simulator S, and thedifference is fed through a shaping filter H and applied, via a switchthat can be used to enable or disable the feedback, to the main signalpath. The low delay corrector LDC provides partial correction for PWMnonlinearity. The predistortion unit provides further correction of PWMnonlinearity. These components will now be explained in more detail.

Lowpass Filtering, Sampling and ADC

The output from the power switches has sharp edges because of the PWMwaveform, and it contains high levels of the switching frequency and itsharmonics.

In making the transition from the continuous-time domain to thediscrete-time domain, the ADC will perform a sampling operation, and alow pass filtering is needed to prevent the switching frequency and itsharmonics from aliasing with the sampling process and corrupting thedigital representation of the output audio signal. This filtering needsto be considered carefully in relation to the sampling frequenciesinvolved and the delay introduced.

Currently, the preferred type of ADC converter for high quality audio isthe high-oversampling type, in which a modulator produces a digitaloutput at, for example, 6 MHz or 12 MHz, which is then decimated to anaudio sampling frequency of typically between 44.1 kHz and 192 kHz.Typically, the decimation takes place in two or more stages, the firststage of decimation producing an output at typically four times thefinal output rate (c.f. section 1.3.2 of 7.) A frequency of 384 kHzwould not be unreasonable for the output of the first stage ofdecimation, which means that the second stage can be dispensed with,which is highly convenient, as it introduces delay.

FIG. 4 shows a preferred form of ADC which consists of an ADC modulatoroperating at a frequency fs_(ADC), for example 6.144 Mhz, followed by asingle stage of decimation to produce an output at fs_(PWM), for example384 kHz. The remainder of the digital circuitry in FIG. 3 also operatesat fs_(PWM), and this is also the PWM switching frequency if class ADmodulation is used.

The PWM output waveform contains the switching frequency and harmoniccomponents having very substantial amplitudes. The fundamental and theharmonics are all modulated nonlinearly by the input to the PWMmodulator. The higher order the harmonic, the more nonlinear is itsmodulation. If the nonlinearly modulated harmonics are aliased down tothe audio band, they will introduce audio distortion into the feedbackchain and hence add distortion to the reproduced audio. Ifaudiophile-grade distortion figures are to be obtained, each frequencythat might alias into the audio band must be attenuated by about 100 dB.

The two processes that might cause aliasing are the sampling atfs_(ADC), and the decimation from fs_(ADC) to fs_(PWM). These processesalias frequency components close to fs_(ADC) and fs_(PWM) respectively,and components close to their respective harmonics, into the audiofrequency band. It is the job of the analog lowpass filter in FIG. 3 toensure that components near fs_(ADC) and its harmonics are sufficientlyattenuated, while it is the job of the decimation filter in FIG. 4 toensure that the components near fs_(PWM) and its harmonics aresufficiently attenuated.

In the case that fs_(ADC) is about 6 MHz, a second order lowpass filterhaving two poles at 40 kHz (4 μs) will provide attenuation of about 87dB at fs_(ADC). A third order filter having three nonresonant poles at200 kHz (0.8 μs) would provide very similar attenuation at fs_(ADC), andwould probably be preferable, but for simplicity we shall base much ofwhat follows on the assumption of a second order filter.

The lowpass filter introduces significant or substantial delay, but ifit is a minimum-phase filter, most of the delay can be compensateddigitally, either in the correction unit LPF⁻¹ or later in the signalchain if LPF⁻¹ is not present. For example, two poles of 4 μs each willapparently produce a group delay of 8 μs near DC, but if the filterLPF⁻¹ is given the response:

LPF ⁻¹=4.36757−4.55540.z ⁻¹⁺1.18783.z ⁻²

at a sampling frequency of 384 kHz, as will be discussed later, then5.6764 μs of delay is recovered, reducing the net delay to 2.3236 μs, or0.89 samples.

Decimation Filter

In a commercial high-oversampling ADC, it is almost universal practiceto perform the first stage of decimation using a cascade of “comb”filters, each of which has a frequency response:

$\begin{matrix}\frac{\sin \left( \frac{\pi \; f}{fs} \right)}{N\; {\sin \left( \frac{\pi \; f}{N\; {fs}} \right)}} & {{Equation}\mspace{14mu} 1}\end{matrix}$

where f is frequency, fs=fs_(PWM) is the output sampling frequency and Nis the decimation ratio fs_(ADC)/fs_(PWM). Comb filters have aparticularly simple implementation (c.f. section 1.3.3 of reference 7below.)

The single comb filter provides infinite attenuation of fs_(PWM) and itsharmonics, but considering distortion at 20 kHz, the critical factor isthe attenuation of components 20 kHz away from a harmonic. Withfs_(PWM)=384 kHz and N=16, the attenuation at (384 kHz−20 kHz)=364 kHzrelative to the response at 20 kHz is 25.15 dB. Hence a cascade of fourcomb filters is required if an attenuation of about 100 dB is required.

The group delay of a single comb filter is just under half a sampleperiod at the output rate. More precisely, it is (N−1)/(2 N)periods=0.46875 periods when N=16. Four combs produce a delay of 2(N−1)/N periods=1.875 periods, or 4.88 μs when fs=384 kHz.

This delay is accompanied by an amplitude droop of 15.63 dB at theNyquist frequency of 192 kHz. An amplitude droop can be corrected bysubsequent filtering at the decimated rate, and if this is done using aminimum-phase filter, the group delay at DC is reduced.

In detail, from result (a) in group I of the “Tabulation of RelationsBetween Real and Imaginary Components of Network Functions”, on page 334of the 1975 edition of reference 2 below, we can deduce that the groupdelay near DC of a discrete-time minimum-phase filter is:

$\begin{matrix}{{delay}_{DC} = {\frac{1}{{fs}^{2}}{\int_{0}^{\frac{fs}{2}}{\frac{\ln \left( \frac{G(0)}{G(f)} \right)}{{\sin \left( \frac{\pi \; f}{fs} \right)}^{2}}\ {f}}}}} & {{Equation}\mspace{14mu} 2}\end{matrix}$

where G(f) is the amplitude gain of the filter. We shall refer to thisequation as the “Bode formula”.

In the case of a droop correction filter, the gain G(f) is generallyhigher than the DC gain G(0), so the logarithm is negative and the groupdelay will also be negative, i.e. the filter produces a phase advance.

The correction filter required to correct a comb droop is:

${G(f)} = \frac{N\; {\sin \left( \frac{\pi \; f}{N\; {fs}} \right)}}{\sin \left( \frac{\pi \; f}{fs} \right)}$

Inserting this G(f) in the Bode formula with N=16, we find a negativedelay: −0.11998 sample periods. Adding the comb delay of 0.46875 periodswe find a net delay of 0.34877 periods.

A cascade of four combs with amplitude droop corrected will thereforeproduce a delay of 4×0.34877=1.39507 periods or 3.633 μs at fs_(PWM)=384kHz. This delay is significant or dominant in the context of thefeedback loop.

We now consider the design of a decimation filter with minimal delay.The purpose of the filter is to attenuate the frequencies that wouldotherwise cause aliased images. A minimum-phase filter will provide aspecified attenuation with minimal delay. In the Bode formula (Equation2,) the term G(0)/G(f) is the attenuation relative to DC, expressed asan amplitude ratio. Increasing the attenuation at any frequency otherthan DC will increase the delay, and attenuation at lower frequencies ismore important than that at higher frequencies because of the term

${\sin \left( \frac{\pi \; f}{fs} \right)}^{2}$

in the denominator. Therefore, the decimation filter should not havemore attenuation than is needed at any given frequency.

The frequencies where attenuation is required most critically are thosethat would alias into the audio band, which is conventionally taken as0-20 kHz. Thus in the case considered above, high attenuation isrequired over the critical frequency ranges 364 kHz-404 kHz, 748 kHz-788kHz, etc. The decimator will alias signal components outside theseranges so that they appear as ultrasonic components in the feedbackpath, from where they are injected into the forward path. The decimationfilter therefore needs to be designed with some regard to its responseoutside the critical frequency ranges, in order to limit the totalultrasonic energy injected into the in the main signal path.Nevertheless, its response can be allowed to rise very substantiallyoutside the critical ranges.

FIG. 5 shows the coefficients of an 80-tap FIR decimation filterdesigned according to these principles, and FIG. 6 shows the attenuationit gives, plotted over the full Nyquist range from 0 to fs_(ADC)/2,which is 3.072 MHz in the case considered. This filter was designedusing a simple linear least-squares procedure. A penalty function wasconstructed giving a very high weight to the mean-square response in thecritical frequency ranges and a moderate weight to the mean-squareresponse elsewhere. The first tap was constrained to be unity, and theremaining 79 taps were chosen by least squares to minimize the penaltyfunction. Then the filter was renormalized to give unity response at DC.It was found empirically that a filter with fewer than 80 taps failed toprovide adequate attenuation over the critical frequency ranges. Longerfilters are in principle better, resulting in the attenuation reducingmore rapidly away from the frequency ranges where it is needed, but theimprovement in overall system performance is small.

FIG. 7 is a comparison between this 80-tap FIR filter and the cascade offour combs described earlier. Both filters provide at least the desired100 dB of attenuation over the critical frequency ranges, but the combfilter provides unnecessarily large attenuation at some frequenciesoutside these ranges, and also unnecessarily large attenuation at theexact multiples of the switching frequency.

Neither filter is completely flat over the output Nyquist frequencyrange 0-192 kHz. It is wasteful to try to achieve flatness over thisrange in a filter that operates at 6.144 MHz, since a similar resultcould be achieved more economically by a postprocessing with aflattening filter running at the output rate of 384 kHz. In the contextof the amplifier, such flattening is folded with other filtering, butfor the purpose of comparison we temporarily suppose that the eachdecimation filter will have its response 0-192 kHz flattened by adedicated filter. It will be apparent that the alias performance is notaffected by such a filter, as raising the response at, say, 50 kHz, willincrease the desired signal and the alias products that fall at 50 kHzin the same ratio. It would be equivalent to pre-filtering (i.e. beforethe decimator) so as to raise the response both at 50 kHz and at all thefrequencies 384 kHz±50 kHz, 768 kHz±50 kHz etc. that will alias to 50kHz. Closer examination shows that this equivalence applies not only tothe alias performance but also to the total delay.

Therefore, a valid way to characterize a decimation filter is to plotthe ratio of its response at a frequency to its response at the image ofthat frequency in the output Nyquist range. This gives the response onthe assumption that the output Nyquist range has been flattened by oneof the two equivalent means described above.

The 80-tap FIR filter and the four cascaded combs are compared on thisbasis in FIG. 8. The curve shapes are now different but once again thecascaded combs are seen to provide substantial unnecessary attenuation.

From the coefficients in FIG. 5, the group delay at DC of the 80-tap FIRfilter is easily computed as 12.94 sample periods at the input rate or0.809 periods at the output rate of 384 kHz. A flattening filterimplemented at fs=384 kHz that flattens the response over 0-192 kHz hasa group delay of −0.1869 sample periods. The group delay of theflattened filter is therefore 0.809+(−0.1869)=0.6221 periods. This is tobe compared with the corresponding figure of 1.39507 periods that wasobtained for the cascade of four combs. The 80-tap FIR filter thus has adelay advantage of greater than a factor two in this instance.

The delay advantage of the FIR filter would have been less if, in theleast-squares procedure described above, the weight given to theresponse outside the critical regions had been reduced. Noise generatedby the ADC is the principal factor determining the how much attenuationis needed outside the critical regions. This will be different from oneADC to another: also it is frequency-dependent so the least-squaresweighting should also be frequency-dependent outside the criticalregion. Weighting within the critical regions should model thesensitivity to aliased sidebands of the switching frequency's harmonics.This will depend on the transfer function of the analog lowpass filter.When all these considerations have been taken into account, theresulting FIR decimation filter may have a greater delay than the oneillustrated. Nevertheless the superiority over cascaded combs is likelyto remain.

A distinction between the prior-art comb filter and the decimationfilter disclosed herein is illustrated in FIG. 9 and FIG. 10. FIG. 9shows the z-plane transfer function zeroes of a single comb according toequation 1 with N=16. There are fifteen zeroes, equally spaced on theunit circle except for the absence of a zero at z=1+0i. Each zero is ata frequency corresponding to the sampling frequency fs=fs_(PWM) or aharmonic thereof. In a cascade of four combs, each zero in FIG. 9becomes a quadruple zero, so again all zeroes are on the unit circle andare at the sampling frequency or a harmonic.

FIG. 10 shows the z-plane transfer function zeroes of the 80-tap FIRdecimation filter described above. It will be seen that most of thezeroes are still close to the unit circle, but there is some spreadingalong the circle in order to give more uniform attenuation over eachcritical range. Further detail of the cluster of five zeroes close toz=0+1i is shown in FIG. 11. It is preferable that all zeroes should lieinside the unit circle as shown, but this may not be achieved inpractice because of coefficient rounding errors.

Feedback Stabilization Using a Simulator

As already explained above, power supply variation and PWM nonlinearitypose significant stability problems to a feedback loop designed usingprior art methods. The present amplifier provides, as shown in FIG. 3, asimulator S that models known aspects of the pulse width modulator andpower switch response. An error signal e is then derived by subtractingthe simulator output from the output of the LPF⁻¹ correction filter, oralternatively from the ADC output if the LPF⁻¹ correction unit is notpresent.

There are thus two paths from the output u of the noise shaper to thesubtractor. The first path is via the simulator S, while the second isvia the pulse width modulator, the output switches, the analog lowpassfilter, the ADC and LPF⁻¹. This second path will be referred to hereinas the measurement path, and for future reference we display themeasurement path in FIG. 12.

The simulator S is intended to model the measurement path. If themodelling were perfect, the two paths would balance and the error signale would be zero. This is the principle of the feedback stabilisation—ifthe error signal is zero, there is no feedback and there can be nooscillation. In practice, the modelling is not perfect. Nevertheless,the tendency to oscillation can be suppressed very considerably. Forexample, supposing that each path had a small-signal gain ofapproximately unity but that, at some frequency, there was a 5%difference between the gains of the two paths, then the gain from pointu to point e in FIG. 3 would be approximately 0.05. Considering now thepath from e back to u, the noise shaper can be assumed to have a gain ofunity; the gain of LDC can be bounded by a number such as 1.7 (c.f. FIG.26), so the shaping filter H would need to have a gain of1/(0.05×1.7)=11.8 approximately for the gain round the loop to exceedunity and hence for an oscillation to be self-sustaining. In practice, avery useful improvement in performance can be obtained while restrictingH to have a gain very much less than this.

Ideally, the simulator should model both the linear response andsubstantial nonlinear aspects of the measurement path. The modelling isrequired to be reasonably accurate over the full Nyquist range 0 tofs_(PWM)/2, and over this range the PWM nonlinearity produced by themodulator is very significant, as has been discussed with reference toFIG. 2.

PWM nonlinearity has a precisely known mathematical form that has beenextensively discussed in the literature (4, 5). FIG. 13 shows a simplemodel of PWM nonlinearity, modelling a symmetrical double-edgemodulator. The input x represents pulse length as a proportion of thePWM switching period. In class AD modulation, x ranges from 0 to 1,while in class BD modulation, x ranges from −1 to +1. The model passesthe signal with a delay of one sample, and in addition generates a cubicnonlinearity followed by a digital approximation to a doubledifferentiation. Except for delay, this model is accurate in the audiorange and could reasonably be used as the basis for the simulator S iffollowed by linear filtering to model the linear response of themeasurement path. The model of FIG. 13 becomes increasingly inaccuratewith increasing frequency, but this may not matter if the feedback loopis not active up to the Nyquist frequency.

Derivation of Low-Rate Simulator

The simulator of FIG. 13 is generic in that it does not presuppose theuse of any particular type of ADC. We now describe a modelling procedurefor use with an oversampling ADC, whereby the resulting simulatoraccurately models not just the PWM modulator, but also the remainingcomponents in the measurement path.

In FIG. 14 the horizontal axis represents time in units of the period ofan oversampling ADC clock, running at, for example, 6.144 MHz. We assumea PWM switching frequency of 384 kHz, and the time axis is conceptuallydivided into frames each containing one PWM pulse and in this case oflength 16 beats of the ADC clock, as shown by the vertical lines. Thefirst frame runs from t=0 to t=16, and the dashed line shows a PWM pulseplaced centrally within that frame, of length 8 beats, i.e. with x=0.5,where x is the pulse length as a proportion of the PWM switching period.The solid line is the response to that pulse of a second-order all-poleanalog lowpass filter having two nonresonant poles of 4 μs each (henceapproximately −6 dB at 40 kHz). This filter response can be calculated,for any value of x, using standard Laplace transform techniques.

The dashed lines in FIG. 15 represent three Dirac pulses at t=0, t=16and t=32 of areas 4.36757, −4.55540 and 1.18783 respectively. If thesolid curve in FIG. 14 is convolved with this pulse sequence, the solidcurve of FIG. 15 results. The three pulse areas sum to unity, and theirheights are calculated by pole-zero matching so that the convolvedimpulse response is finite: with x=0.5 the solid curve in FIG. 15 iszero for t≧44; with x=1 the PWM pulse would be four units longer in eachdirection and the convolved impulse response would extend in time fromt=0 to t=48.

If the convolved response of FIG. 15 is further convolved with thedecimation filter of FIG. 5, the response shown in FIG. 16 results. Thisis zero from t=123 onwards, or with x=1 it would be nonzero over therange 0<t<127. We will denote this response by d(x, t).

If we now sample d(x, t) at t=0, 16, 32, . . . , we will obtain thesampled sequence:

-   -   0, d(x, 16), d(x, 32), d(x, 48), d(x, 64), d(x, 80), d(x, 96),        d(x, 112), 0, 0 . . .        sampled at the PWM switching frequency of 384 kHz.

We now itemize the processing steps described above. In order, they are:

-   -   convolution with lowpass filter    -   convolution with three-pulse sequence    -   convolution with decimation filter    -   sampling at fs_(PWM)

The three convolutions are conceptually continuous-time convolutions,the decimation filter being regarded as a sequence of Dirac deltafunctions for this purpose. The person skilled in the art willappreciate that the above steps are equivalent to the following:

-   -   convolution with lowpass filter    -   sampling at fs_(ADC)    -   sampled convolution with decimation filter    -   decimation from fs_(ADC) to fs_(PWM)    -   sampled convolution with three-pulse sequence        wherein the decimation filter and the three-pulse sequence are        now conventional FIR digital filters This processing sequence        can be identified with the processing shown in FIG. 12, if LPF⁻¹        in FIG. 12 is an FIR filter that performs the convolution of the        three-pulse sequence:

LPF ⁻¹=4.36757−4.55540.z ⁻¹+1.18783.z ⁻²

In making this identification, we assume that the ADC modulator passesinput to its output without change. We also assume that the effect ofthe power switch is to give each PWM pulse a height proportional to thepower supply voltage V_(CC), this being assumed not to change during thepulse.

It follows that the processing of FIG. 12 transforms an input value xinto an output pulse sequence 0, d(x, 16)×V_(CC), d(x, 32)×V_(CC), . . ., d(x, 112)×V_(CC), 0, 0 . . . . All processing after the pulse widthmodulator is linear, so superposition applies and it follows that themeasurement path of FIG. 12 can be modeled by the architecture of FIG.17, in which each of the d(x, 16), d(x, 32) etc. is represented by anonlinear function generator with input x.

To produce a practical simulator, we approximate each the nonlinearfunction by a polynomial in x. For the case discussed above, eachfunction d(x, .) was evaluated at thirty values of x (x=1/32, 2/32, . .. , 30/32) and the following coefficients obtained by a linearleast-squares procedure:

d(x, 16)=0.0819616x+0.040866x ³+0.0011542x ⁵

d(x, 32)=0.8066498x−0.0849205x ³−0.0056800x ⁵

d(x, 48)=0.0438820x+0.0533158x ³+0.0114638x ⁵

d(x, 64)=0.0784520x−0.0229539x ³−0.0122444x ⁵

d(x, 80)=−0.1146495x+0.0269429x ³+0.0073815x ⁵

d(x, 96)=0.0921093x−0.01881471x ³−0.0024272x ⁵

d(x, 112)=0.115946x+0.005619x ³+0.0003500x ⁵

wherein the approximation error is typically less than 10⁻⁵. Using thisapproximation, the model of FIG. 17 can be implemented on a morepractical architecture such as that shown in FIG. 18.

The generalized simulator of FIG. 18 implements even powers as well asodd powers of x. The coefficients SIMpbxy can be programmed to reflectthe decimator and filtering used for the feedback. This allowsstraightforward treatment of class AD modulation, where it is convenientto re-define x so that pulse length is a proportion (1+x)/2 of the PWMswitching period, and x is zero in the no-signal condition. “PSe” inFIG. 18 represents an estimate of V_(CC). “NS” in FIG. 18 is to beidentified with the noise shaper of FIG. 3, thus FIG. 18 is able toproduce an output whose linear term is being noise shaped but whosenonlinear terms are derived from the signal before noise shaping.

It is desirable that the simulator model the behavior of the measurementpath under overload conditions as well as during normal operation. Sincethe pulse length cannot exceed 100% of the repetition period, it isdesirable that x be limited so that |x|≦1. If, in the amplifier of FIG.3, the noise shaper incorporates internal clipping, this can be arrangedautomatically. If not, or if part of the input to the simulator is takenprior to noise-shaping as envisaged in FIG. 18, then a signal limitingdevice may be inserted immediately after LDC in FIG. 3 in order toenforce the condition |x|≦1.

The filter LPF⁻¹ displayed above has two zeroes in the z-domain. Forreasons to be explained, its order and coefficients are chosen so thatits zeroes cancel the s-domain poles of the analog lowpass filter.Specifically, the lowpass filter has two coincident poles at s=0.25μs⁻¹, from which the zeroes of LPF⁻¹ are calculated as:

z=exp(−s.τ)=exp(−0.25×1 MHz/384 kHz)=exp(−0.65104)=0.5215

Hence:

LPF ⁻¹=(z−0.5215)² /z ²/(1−0.5215)²=4.36757−4.55540.z ⁻¹+1.18783.z ⁻²

More generally, the lowpass filter may be of higher order and/or havecomplex poles, in which case LPF⁻¹ will be of higher order and/or havecomplex zeroes, again calculated by pole-zero matching.

Sampling Point and Computational Delay

Comparing the conceptual simulator of FIG. 17 with the polynomialapproximations above for d(x, .), we can ignore nonlinear terms andderive small-signal response of the simulator (ignoring themultiplication by V_(CC)) as:

0.0819616.z⁻¹+0.8066498.z⁻²+0.0438820.z⁻³+0.0784520.z⁻⁴−0.1146495.z⁻⁵+0.0921093.z⁻⁶+0.0115946.z⁻⁷

The common factor z⁻¹ makes possible the inclusion of the simulatorwithin a loop such as that of FIG. 3 that does not have other delayelements, without creating a delay-free feedback loop.

The z⁻¹ delay can be understood also with reference to FIG. 14. Weidentify samples at the switching frequency fs_(PWM) with the timeinstants t=0, 16, 32, 48 etc. in FIG. 14. An input to the pulse widthmodulator at t=0 is able to modulate the pulse centred on t=8, and thishas an influence on the waveform in FIG. 16 at the next sampling instantt=16. There is thus a delay of precisely one fs_(PWM) sample before thefirst response of the measurement path is seen. Ideally, this responsewould be fed round the feedback loop of FIG. 3 and would influence thewidth of the next PWM pulse (not shown) that is centred on t=24.

This ideal situation is hard to realize in practice as x becomes closeto 1 and the pulse edges become closer to the sampling points. Any delayin the pulse width modulator, or computational delay in the signal paththat feeds it, will make it impossible for a signal received at t=16 tocontrol a symmetrical pulse centred on t=24 whose leading edgeapproaches t=16. Conversely, any delay in the measurement path will meanthat the trailing edge of the pulse centred on t=8 is not able toinfluence the sample at t=16 to the intended extent.

One way to address this problem is insert an extra sample of delay inthe feedback loop, so that the measurement of the pulse that is centredon t=8 does not influence the pulse that is centred on t=24 but insteadhas its first effect on the pulse that is centred on t=40. This createstiming slack so that computational delay can be accommodated and alsoallows the sampling points to be moved slightly later in time to accountfor any delay in the measurement path. The simulator is given anadditional factor z⁻¹ so that its small-signal response (ignoring themultiplication by V_(CC)) is:

0.0819616.z⁻²+0.8066498.z⁻³+0.0438820.z⁻⁴+0.0784520.z⁻⁵−0.1146495.z⁻⁶+0.0921093.z⁻⁷+0.0115946.z⁻⁸

However, this solution is non-optimal because the additional factor z⁻¹in the feedback loop reduces its effectiveness in correcting errors.

An alternative is perform the sampling at some point within the PWMframe, i.e. between t=0 and t=16 in FIG. 14, so that the next pulse canbe influenced in response to the sample even at full modulation (x=1)and allowing for computational delay. One scheme is to sample at thepulse centres t=8, t=24, t=40 etc. or just slightly afterwards to allowfor delay in the measurement path. If using class AD modulation, this ismathematically attractive because modulation will not cause a pulse edgeto cross a sampling point, so all the d(x, .) functions will becontinuous and analytic, leading to good approximation by a low-orderpolynomial.

Another scheme is to sample later in the PWM frame, but still allowingsufficient time for the sample value to influence the next pulse,allowing for computational etc. delays. In this scheme, there will be aqualitative change in behavior when the value of x is such as to causethe trailing edge of the modulated pulse to cross the sampling point,and it will not be possible to approximate the d(x, .) functions asaccurately using low order polynomials. Deeper analysis reveals,however, that if the decimation filter has been designed properly,distortion caused by inaccuracy of the polynomial approximation can besubstantially confined to the ultrasonic region. In order to make thishappen, the polynomials approximating the d(x, .) functions may need tobe optimized jointly rather than individually, with a frequency domainweighting applied to the error criterion.

High-Rate Simulator

There are many architectures that can be used to model the measurementpath. An alternative simulator that is intended for use with anoversampling ADC, and in which the simulation is performed at the higherclock frequency fs_(ADC) is therefore described below.

In FIG. 19, the signals u and e are the input to the PWM modulator andthe feedback error signal, and are to be identified with thecorresponding signals u and e in FIG. 3. Thus, FIG. 19 provides areplacement for the right-hand half of FIG. 3.

In FIG. 19, the pulse width modulator receives an input signal u andproduces an analog PWM waveform that drives the output power switches.The output from the power switches is filtered by an analog lowpassfilter and feeds an ADC modulator operating at a high oversamplingfrequency fs_(ADC) of, for example, 6.144 MHz. The output of themodulator feeds a digital decimation filter before being downsampled bya factor of, for example, n=16, to produce an output at a frequencyfs_(PWM). A digital filter LPF⁻¹ substantially compensates, within theNyquist range, the effect of the analog lowpass filter.

The measurement path in FIG. 19 is thus identical to that of FIG. 3 ifthe unit marked “ADC” in FIG. 3 is expanded as shown in FIG. 4.

The input u to the pulse width modulator in FIG. 19 is fed also to anedge timing determination unit, which derives the timings of the leadingand trailing edge of the pulse that will be produced by the pulse widthmodulator. This information is passed to the antialiassed sampler whichprovides a sampled representation, at sampling frequency fs_(ADC), ofthe pulse that will be produced by the pulse width modulator. Theantialiassed sampler's output sequence is then filtered by a digitallowpass filter before being subtracted from the output of the ADCmodulator.

The edge timing determination unit, the antialiassed sampler and thedigital lowpass filter together form a simulator as shown, thefunctional difference from the simulator S FIG. 3 being that its outputis provided at the higher sampling rate fs_(ADC) and is thereforesubtracted from the measurement path before decimation, rather thanafterwards.

Considering now the antialiassed sampler in more detail, if the pulseedge timings were quantized to beats of the fs_(ADC) clock, thesampler's task would be trivial: for example it would emit the sequence:

-   -   −1, −1, −1, −1, +1, +1, +1, +1, +1, +1, +1, −1, −1, −1, −1        to represent the pulse of length eight clock beats that has been        discussed with reference to FIG. 14. A simple method to        represent a pulse whose length is not so quantized is to use        linear interpolation. For example, a pulse of length 8.4 units        with rising edge at t=3.8 and falling edge at t=12.2 could be        represented by the sequence:    -   −1, −1, −1, −0.6, +1, +1, +1, +1, +1, +1, +1, −0.6, −1, −1, −1

This ‘linear interpolation’ method is equivalent to convolving thecontinuous-time PWM pulse with a narrow rectangular pulse of width onefs_(ADC) clock before sampling at frequency fs_(ADC). The person skilledin the art will be aware that it would alternatively be possible toconvolve with a B-spline having a knot spacing of one fs_(ADC) clock,for better rejection of alias products, and that many otherinterpolation possibilities exist.

The antialiassed sampler also multiplies its output sequence by adigital estimate of V_(CC), in order to model the effective analogmultiplication performed by the output switches.

The digital lowpass filter in FIG. 19 is intended primarily to mimic theeffect of the analog lowpass filter. For example if the analog filterwere an all-pole filter, then the digital filter could also be all-polewhere a pole at s=s_(p) in the analog filter is matched by a pole atz=z_(p)=exp(τ.s_(p)) in the digital filter, where τ=1/fs_(ADC). Thesmall-signal transfer functions of the ADC and of the output switches,if they differ from unity, can also be folded into this filter. Puredelay in the drive circuitry need not appear in the filter, as it can beaccounted for in the edge timing determination unit.

The response of such a digital filter can differ significantly near theNyquist frequency from the response of the analog filter. The differencemay be reduced very substantially by using B-spline convolution in theantialiassed sampler, where the order of the B-spline is one less thanthe order of the analog filter, for example a quadratic B-spline wouldbe used with a third order analog filter.

The digital lowpass filter may be further adjusted to model delay andother non-idealities in the small-signal transfer function of the powerswitches and the ADC modulator.

The pulse width modulator in FIG. 19 may be able to furnish the edgetimings required by the antialiassed sampler, and if this is the casethe edge timing determination unit will not appear as a dedicatedcomponent of the simulator: the simulator will receive its input fromthe pulse width modulator in this case.

In the context of the amplifier of FIG. 3, the filter LPF⁻¹ in FIG. 19can be omitted if the high rate simulator is used. In this case, withinthe design procedure for H that we shall now describe, the transferfunction “LPF⁻¹” should be replaced by unity.

Feedback Loop Filter

With reference to FIG. 3, the features described so far are directedtowards minimising the signal delay through the measurement path (FIG.12), and also towards ensuring that the simulator in FIG. 3 accuratelymodels the measurement path so that the transfer function from u to e isclose to zero over the full Nyquist range 0−fs_(PWM)/2.

If we assume that the transfer function from u to e is actually zero,then the effect of the feedback is easily computed since recirculationdoes not need to be considered. The feedback will multiply the effect ofa disturbance in the power switches by a transfer function

NTF=1−H.P  Equation 3

where H is the small signal transfer function of the feedback filter,and P is the combined small signal transfer function of LDC, the noiseshaper, the pulse width modulator, the power switches, the analoglowpass filter, the ADC and LPF⁻¹. Thus, H.P is the combined transferfunction of all the components, other than the simulator, that form thefeedback loop in FIG. 3

For complete suppression of power switch errors, we would require NTF=0,hence H.P=1, hence H=P⁻¹. That is unlikely to be possible, because it isunlikely that P will have a causal inverse. To make further progress, wedecompose P as

P=M.A  Equation 4

where M is minimum-phase, and A is allpass.

The likely contributors to M are the decimation filter; the analoglowpass filter (which, however, is partially compensated by LPF⁻¹,)droop from the pulse width modulator (c.f. FIG. 2 which, however, may bepartly compensated by the correction unit LDC.) In addition, M includesthe gain of the power switches, which may vary if the power supplyvaries.

Contributors to A include the intrinsic delay of the pulse widthmodulator, propagation delay through the analog and digital electronics,and computational delay. These factors impact A through their effect onthe choice of sampling point as already discussed. The decimation filtermay also contribute to A, for although it is designed as a minimum-phasefilter at its sampling rate of fs_(ADC), its effect when viewed througha sampling process at fs_(PWM) is not necessarily minimum phase. Asimilar consideration applies to the analog lowpass filter.

Only the minimum-phase component of P is causally invertible. If wechoose

H=M⁻¹  Equation 5

then substituting equations (5) and (4) in equation (3):

NTF=1−H.P=1−H.M.A=1−A

At DC, A=1 so the feedback will suppress very low frequency errorsalmost perfectly.

In the example configuration discussed above, with decimation filter asgiven in FIG. 5, second order lowpass filter with matched LPF⁻¹, andwith the computational delay problem addressed by inserting anadditional sample of delay in the feedback loop, we find that, A has agroup delay at low frequencies of approximately 2.8 samples. This delaycorresponds to a phase shift of 0.92 radians at 20 kHz when fs_(PWM)=384kHz, whence |NTF|=0.88, i.e. errors are reduced by 1.1 dB at 20 kHz, or12.8 dB at 5 kHz.

Although A is not causally invertible, it is possible to design aprediction filter H′ that can substantially compensate the phaseresponse of A over an operating frequency range that is less than thefull Nyquist range 0 to fs_(PWM)/2. Study of equation (2) reveals that aminimum-phase filter having an amplitude response that increases abovethe operating frequency range, as sketched in FIG. 20, will havenegative group delay near DC. Given such a filter H′, if we now set

H=M⁻¹.H′  Equation 6

then we find:

NTF=1−A.H′

which gives the possibility of a smaller NTF, over the operatingfrequency range, than results when H is chosen using equation (5).

FIG. 20 shows an H′ that rises above the operating frequency range butfalls at still higher frequencies. The attenuation at higher frequenciesmay be necessary to prevent too much high frequency noise from the ADCfrom being injected through H into the main signal path.

To implement these principles, it is not necessary to design the twofilters M⁻¹ and H′ separately and then combine them. FIG. 21 tabulatesthe coefficients of a feedback filter H designed as a single filter withcoefficients chosen by a linear least-squares optimization procedure,and FIG. 22 shows its amplitude response.

The optimization attempts to balance several criteria. Firstly, in orderto provide maximum feedback advantage in the audio range, it attempts tominimize |1−H.P| evaluated at several frequencies over the range 0-20kHz, with the greatest weight being given to low frequencies. Secondly,the optimization attempts to minimize |H.LPF⁻¹.N_(ADC)| over the fullNyquist range, where N_(ADC) is an estimate of the noise spectrumproduced by the ADC, in order to control noise injection. Thirdly, thereis some penalty attached to the response in the region of its maximum,in this case 39 kHz, in order to control the maximum gain |H.P| andhence provide some stability margin in the event that the simulator andmeasurement paths are not perfectly matched.

FIG. 23 and FIG. 24 provide further detail of the loop characteristicsresulting from the loop filter H just described. FIG. 23 plots |H.P| ona decibel scale. This is the loop gain that would be obtained if thesimulator were removed. Denoting the simulator transfer function by S,the loop gain through H with the simulator present is |H. (P−S)|. Asufficient condition for stability is that |H. (P−S)|<1 at allfrequencies. Rearranging this condition as:

|(P−S)/P|<1/(H.P)

it follows that 1/(H.P) is an estimate of the proportional deviation ofP from S that can be tolerated before stability ceases to be guaranteed.In FIG. 23, |H.P| peaks at about +20 dB, so P and S need to be matchedwithin about 10% in order for the above condition stability to be met.

FIG. 23 does not differ greatly from FIG. 22 in this case. This isbecause the decimation filter has a substantially flat amplituderesponse in the frequency range considered, and the analog lowpassfilter is substantially compensated by LPF⁻¹, so |P| differs onlyslightly from unity.

FIG. 24 is a decibel plot of |1−H.P|, also known as a Noise TransferFunction or NTF. From this, we deduce that errors in the audio range0-20 kHz are reduced substantially, errors in the ultrasonic region 20kHz-90 kHz approximately are increased by up to 20 dB, and errors above90 kHz are not significantly affected. At 20 kHz, errors are reduced by9.3 dB, to be compared with the estimate of 1.1 dB given previously forthe H of equation 5 that does not incorporate prediction.

In practice, it may be preferred to limit the maximum gain of H muchmore severely than shown in FIG. 22 in order to provide stability thatis more robust with respect to differences between P and S, and also tolimit the maximum error amplification in the ultrasonic region. Theseadvantages are at the expense of a smaller reduction of errors in therange 0-20 kHz.

The transfer function P includes the response of the decimation filter,which has so far been assumed to be approximately flat over thefrequency range 0−fs_(PWM)/2. However, there is freedom to adjust thisresponse, and with suitable adjustment, M (equation 4) may be close tounity so the choice H=M⁻¹ would result in an H that is also close tounity, in which case the filter H may be omitted entirely.

Low Delay Correction Unit LDC

The filter H allows the small-signal loop transfer function to beadjusted to achieve the desired compromise between overall stability andeffectiveness of feedback over the operating frequency range of, forexample, 0-20 kHz. However, the transfer function of the pulse widthmodulator varies in response to large signals, as shown in FIG. 2.Depending on the desired amount of feedback, this variation may besignificant in reducing the feedback at high signal excursions, and itis the function of the optional correction unit LDC to provide partialcompensation for this effect. A design of LDC adapted to double-edgedpulse width modulation is shown in FIG. 25.

In FIG. 25, the signal y is fed through a nonlinear function generatorto form P(y), which is subtracted from y to furnish output signal x andalso fed back though a filter with transfer function5/2.z⁻¹−2.z⁻²+z⁻³/2. The LDC unit feeds the pulse width modulator inFIG. 3, and the assumed scaling for x is that x=0 corresponds to a pulseof length zero length while x=1 corresponds to a pulse of length 100% ofthe switching period (1/fs_(PWM)).

Based on the model in FIG. 13, It can be shown that:

P(y)=y ³/12

sufficiently models the non-linearity, while perhaps slightly better is:

P(y)=0.08251487120y ³−0.01495088616y ⁵

The clip unit in FIG. 25 is provided to prevent out-of-range inputsignals from sending the feedback loop contained within FIG. 25 intopersistent oscillation. A clipper operating at levels of ±0.125 would besatisfactory in the position shown.

The small signal amplitude response of FIG. 25, with P(y)=y³/12, isplotted for pulse lengths equal to 0%, 50% and 100% of the switchingperiod in FIG. 26. In each case, the response is minimum phase. Whencombined with the PWM nonlinearity of FIG. 2, the result is flat overthe operating frequency range 0-20 kHz to a high degree of accuracy. Forpulse lengths other than zero, the response of LDC must start to riseover this range to combat the droop shown in FIG. 2. However, forsymmetrical double-edge modulation, the nonlinearity of FIG. 2 isphaseless, so the rising amplitude response must not be accompanied by aphase advance. The constant phase response is achieved to a reasonabledegree of accuracy by making the amplitude response fall at higherfrequencies, as shown in FIG. 26.

In addition to maintaining feedback effectiveness at high signal levels,the LDC unit provides another benefit. Without the LDC unit, ultrasonicnoise originating from the ADC and injected via H into the main signalpath, could intermodulate with itself in the nonlinearity of the pulsewidth modulator and produce intermodulation products within the audioband. This phenomenon has been discussed in 4 in relation to noiseproduced by a noise shaper. The corrector of FIG. 25 substantiallycompensates, within the operating frequency range, products generated bywideband or aliased ADC noise intermodulating with itself. The correctoris unable to correct products generated by intermodulation of noise fromthe noise shaper shown in FIG. 3.

Predistortion

In the prior art, feedback usually attempts to improve the linearity ofa device such as an amplifier. However, feedback according to thepresent invention does not try directly to impose linear behavior, butrather to reduce the deviation from the behavior of a simulator. Moreprecisely, referring to FIG. 3, the feedback attempts to achieve:

m_(feedback enabled)=s_(feedback disabled)  Equation 7

over the operating frequency range, where m is the output of themeasurement path and s is the output of the simulator. In the discussionthat follows, we shall assume that equation 7 holds exactly.

The predistortion unit in FIG. 3 receives an input signal i and isdesigned to invert nonlinear aspects of the combination of LDC and S. Itis assumed that the noise shaper can be disregarded in a large-signalanalysis. Thus, the predistortion unit provides a corrected signal csuch that, over the operating frequency range and with feedbackdisabled, the simulator output s closely follows a linearly filteredversion of i. Hence by equation 7, when feedback is enabled, m willclosely follow a linearly filtered version of i.

Of more direct interest, however, is the output o of the amplifier,which we assume to be in linear dependence on the output p of the powerswitches. We would therefore like to know that p is linearly related tothe amplifier input i.

The path from p to m, comprising the analog lowpass filter, ADC andLPF⁻¹ unit, should be substantially linear. The path contains a samplingprocess, but the decimation filter should ensure that, over theoperating frequency range, m is not significantly contaminated by aliasproducts. Therefore, m should be linearly related to p over theoperating frequency range. Consequently, if the combined effect offeedback and predistortion is to ensure that m is linearly related to i,it follows that p is linearly related to i. Thus, the amplifier as awhole is linear over the operating frequency range, as required.

Depending on the design of the predistortion unit, the small signaltransfer function from i to p may or may not be a pure delay. If it isnot pure delay, it can be compensated by a linear compensator placedprior to the predistortion unit. If desired, correction for the transferfunction of the LC filter can also be applied at this point.

There are several ways to design a predistortion unit. One is to use thetechniques of nonlinear system identification to derive a Volterraseries expansion of the nonlinear system, and then to invert theVolterra series. A method that does not require advanced mathematics wasproposed by Gerzon (reference 1 below). First-order Gerzon correction isshown in FIG. 27. Gerzon's method corrects nonlinearity in a weaklynonlinear system whose small signal transfer function approximates apure delay τ. The element N to the right of the dashed line in FIG. 27represents the nonlinear system to be corrected. Gerzon's methodrequires one or more replicas of the nonlinearity to be corrected, henceanother element N in the predistortion unit shown to the left of thedashed line. The first-order predistortion unit multiplies the inputsignal by two and delays by τ, then subtracts the distorted signalfurnished by the replica nonlinear element N. The result of thesubtraction is a predistorted signal. If the predistorted signal is fedto another nonlinear element N, the resulting output will, subject toconditions on N, contain substantially less distortion than if the inputsignal were fed to N directly.

Gerzon's method may be nested. That is, the total system of FIG. 27 canitself be considered as a nonlinear element that can be compensated inthe same way. Gerzon (reference 1) describes other methods by which thehigher-order correction can be obtained.

Referring again to FIG. 3, if it is desired to make the simulator output(with feedback disabled) follow the input signal, then the predistortionunit must apply compensation for the cascaded combination of the LDCunit and the simulator S. (The noise shaper in FIG. 3 is assumed to havea unity transfer function and to be modelled by additive noise, whichcannot be compensated.) If the element N in FIG. 27 is replaced by thecascade of LDC and S, we obtain FIG. 28, which is intended to be areplacement for the predistortion unit in FIG. 3.

The small signal transfer function of S is, in general, not completelyflat in amplitude, nor linear in phase. It does not necessarilyapproximate a pure delay τ to the accuracy required for the Gerzoncorrection of FIG. 28 to be optimally effective. An improvement is toprecede the cascade of LDC and S by a linear correction unit S_(lin) ⁻¹that substantially corrects the amplitude response of S and linearizesits phase response, so that S_(lin) ⁻¹.S approximates a pure delay atleast over the operating frequency range. Thus, each N in FIG. 27 isreplaced by (S_(lin) ⁻¹.LDC.S). As there are two instances of N in FIG.27, there are two instances of S_(lin) ⁻¹ when the replacement is made.If S_(lin) ⁻¹ is minimum phase, then each instance of LDC can bepreceded immediately by an instance of S_(lin) ⁻¹. However, linearisingthe phase response of S may require that S_(lin) ⁻¹ have allpassfactors, and if this is the case, S_(lin) ⁻¹ should not appear insidethe feedback loop. Therefore, both instances of S_(lin) ⁻¹ are placed inthe predistortion unit, as shown in FIG. 29. FIG. 29 in its entirety maybe used as the predistortion unit shown in FIG. 3.

It is possible to derive predistortion independently of the simulator S,if a suitable model of the forward path from u to p is available. Forexample, if the output switches themselves are considered perfect, thepath from u to p contains only the pulse width modulator, for which avariety of low-frequency models can be constructed, one of which hasbeen shown in FIG. 13. Moreover, if the output switches are perfect,then the simulator S should model the measurement path very accurately,and the feedback signal f should be essentially zero. Under thesecircumstances, the path from i to p can be linearized by a predistortionunit that compensates the path from d to p.

Two alternative predistortion units adapted to compensate the path fromd to p, i.e. the cascaded combination of LDC and the pulse widthmodulator, are shown in outline in FIG. 30 and FIG. 31. In FIG. 30, theGerzon principle of FIG. 27 is applied directly to the cascadedcombination of LDC and a model of the pulse width modulator. In FIG. 31,the Gerzon principle is applied to the pulse width modulator only,leaving LDC to be compensated by a separate unit LDC⁻¹.

FIG. 32 shows more detail of an example predistortion unit according tothe principle of FIG. 31, i.e. using a separate compensator for LDC. Thepart of FIG. 32 to the left of the dotted line is derived from thecorresponding part of FIG. 31, using the PWM model of FIG. 13.

The part of FIG. 32 to the right of the dotted line is a separatelyderived inverse of the LDC of FIG. 25. It is an exact inverse (up to theclip point) if the nonlinear function Q(x) satisfies:

Q(x)=P(x+Q(x))

Taking the simpler choice for P:

P(y)=y ³/12

this reduces to:

Q(x)=(x+Q(x))³/12

There are various ways to approximate the solution to this equation tofacilitate real-time computation. If only moderate accuracy is required,it may be adequate to take one or more terms of the power seriesexpansion:

${Q(x)} = {{\frac{1}{12}x^{3}} + {\frac{1}{48}x^{5}} + {\frac{1}{144}x^{7}} + {\frac{55}{20736}x^{9}} + \ldots}$

Similar methods can be used to furnish Q(x) for other choices of P.

FIG. 33 shows more detail of an example predistortion unit according tothe principle of FIG. 30. The area labelled “ZDCORR” in FIG. 33 is to beidentified with FIG. 25, “Pzd” being the polynomial P(y) discussedabove. The area labelled “PWMCORR” implements a fifth order model of PWMnonlinearity that is conceptually similar to the third order model ofFIG. 13. For implementation convenience, whereas FIG. 13 implements asingle nonlinearity followed by a polynomial in z⁻¹, in FIG. 33 theterms have been grouped according to powers of z⁻¹ to form thepolynomial nonlinearities P50 through P54. The dominant term would be alinear term in the polynomial P52, which is fed from a cascade of twoz⁻¹ delay elements. This term has been removed from P52 and is accountedfor instead by the path from the output of ZDCORR to the summation nodeat the top of FIG. 33. This transformation reduces the sensitivity ofthe final output to noise introduced by the quantizer Q that feeds thePWMCORR block, and thus allows a coarser quantizer to be used, forimproved efficiency in implementing the polynomial nonlinearities P50through P54.

Adaptation to a Varying Power Supply

Sometimes, a PWM amplifier is required to operate off an unstabilizedpower supply, in which case the supply voltage may deviate from itsnominal value by 10% or more. With all commonly used output switchconfigurations, power supply variation causes variation of the forwardgain. With some configurations it will also cause a varying DC offset,but here we shall consider gain variations only.

FIG. 34 shows an amplifier similar to the amplifier in FIG. 3 but withimprovements directed towards maintaining performance when the powersupply varies. A power supply estimation unit provides an estimatePS_(e) of the gain of the output stage taking into account the powersupply, and also its inverse PS_(e) ⁻¹. It is convenient to assume thatPS_(e) is unity when the power supply voltage has its nominal value.

There are several ways in which the power supply estimation unit mayoperate. One would be to continuously characterize the gain of theamplifier, while another would be to continuously measure the powersupply voltage, for example using an ADC.

As described previously with reference to FIG. 12 through FIG. 18, thesimulator is designed to model the measurement path, and in particularits gain. FIG. 34 assumes a simulator having a fixed gain, followed by amultiplication by PS_(e). An alternative, and better, procedure is touse a simulator as shown in FIG. 17 or FIG. 18 that provides explicitmodeling of the effect of V_(CC) or PS_(e).

The feedback filter H is derived in dependence on the simulator, and asdiscussed previously, H substantially inverts the response of the filterM, defined by equation (4), which includes the gain of the powerswitches. In practice, it is more convenient to have a filter H thatdoes not vary dynamically, but is followed by a multiplication by PS_(e)¹, as shown in FIG. 34.

The elements described so far ensure that the feedback loop continues tooperate correctly if the power supply voltage changes, but there remainsa change in the gain of the forward path of the amplifier, which is notcorrected. This may be corrected by multiplying the input signal byPS_(e) ⁻¹, prior to the predistortion unit, as shown in FIG. 34.

Reference Path

The invention as described so far applies feedback to an amplifier viaan ADC. The ADC has to handle the full dynamic range of the amplifier'soutput, hence the noise and distortion performance of the amplifier isunlikely to be better than that of the ADC.

It may or may not be convenient to incorporate an ADC of adequately highperformance. The prior art feedback design described in U.S. Pat. No.6,373,334 allows an ADC of reduced performance to be used. In this priorart design, the ADC is fed with the difference between a scaled versionof the power switch output and a low-level PWM waveform produced byreference switches that are driven from the same pulse width modulatoras the power switches. It is the design intention that this differencebe small, thus reducing the dynamic range that has to be handled by theADC. However, the difference is likely to increase if the gain of thepower switches changes because of power supply variation.

FIG. 35 shows an amplifier incorporating a reference switch that isdriven from a separate pulse width modulator, identified as “Pulse widthmodulator ref”. The signal path along the top of the diagram is likethat of the prior art amplifier of FIG. 1, save that the referenceswitch is a precision switch operating a low power level, and the outputfilter components L and C are omitted. Therefore, the reference outputsignal r will follow the input signal i to high precision, within theoperating frequency range, be a high-precision replica of the inputsignal i. In particular, the gain from i to r can be assumed constant,since the reference switch will use a local stabilized voltage referenceV_(ref) (not shown), and thereby be immune from variations in V_(CC).

In FIG. 35, the analog lowpass filter is shown as having a differentialinput. In practice, there will be resistive attenuation (not shown)between the output p of the power switches and the positive input of thefilter. Another possibility is to invert digitally the drive to thereference switch, so that the signals p and r can be added rather thansubtracted. Passive resistive summation can then be used prior to afilter having a single-ended input.

The design intention is that the signals from p and r shouldsubstantially cancel when they are combined, thereby substantiallyreducing the dynamic range requirement on the ADC. If passive resistivesummation is used, the dynamic range handled by the lowpass filter isalso reduced.

Also provided in FIG. 35 is a second simulator identified as “S_(ref)”.This simulator is intended to model, apart from the sign inversion, thesignal path from point u_(ref) to point m, just as the simulator“S_(main)” models the path from u_(main) to m. Therefore, taking intoaccount both the path through the reference switch, and the path throughS_(ref), the gain from point u_(ref) to point e should ideally be zero.Thus comparing FIG. 35 with FIG. 34, the addition of the referenceswitch and its simulator to the amplifier of FIG. 34 should ideally notaffect the total performance save that the ADC in FIG. 35 is required tohandle a smaller signal than the ADC in FIG. 34.

It is, however, necessary to be aware that, although the signals p and rin FIG. 35 may cancel substantially at audio frequencies, high frequencyoverload also needs to be considered. The signals at p and r are bothPWM waveforms, but they do not in general have the same mark:spaceratio. For example, the mark:space ratio of the waveform at p will varyin response to variations of V_(CC) whereas the waveform at r isindependent of V_(CC). Therefore, even if the drive to the referenceswitch is inverted and passive resistive summation of the signals at pand r is used to provide a substantially zero result at low frequencies,there will remain transient fast edges that may need to be filteredpassively before they reach active devices in the analog lowpass filter.Further, each of the signals r and p has a substantial component atfs_(PWM). It will be advantageous to provide delay adjustment in the twopulse width modulators so that these two components at fs_(PWM) can besubstantially aligned in phase. Even if they are so aligned, however,the differing mark:space ratios will cause different amplitudes of thetwo components, so cancellation at fs_(PWM) cannot be exact. Thisconsideration may present a requirement for significant attenuation atfs_(PWM) in the analog lowpass filter, taking into account thecapability of the ADC to handle high frequencies.

The “Predistortion main” unit in FIG. 35 is not intended to be afunctional replica of the “Predistortion ref” unit, because of thepresence of LDC.

FIG. 35 provides for substantial cancellation between the signals thatfeed the ADC during normal operation, but if the amplifier clips, it isnot guaranteed that the main and the reference paths will clipsimultaneously, in which case larger signals can appear at the ADC. Thisproblem is addressed in FIG. 36, which shows a detail of the amplifierof FIG. 35, but with the addition of a clip processor which has twooutputs and provides two adjustment signals, which are then added to thetwo pre-clip signals pec_(ref) and pec_(main) to produce the post-clipsignals poc_(ref) and poc_(main). In normal operation, the twoadjustment signals are zero. If either of the two pre-clip signalsexceeds the valid signal range handled by the pulse width modulators,for example −1 to +1, then the clip processor must emit an adjustmentsignal of opposite sign. For example, if pec_(main) has the value 1.2,an adjustment signal of −0.2 added to the main path would givepoc_(main)=1.0. If the reference path were not similarly modified, thisperturbation would result in an uncanceled perturbation at the input tothe ADC. Hence, the clip processor emits also an adjustment to thereference path. If the power supply voltage has deviated from itsnominal voltage, this has a multiplicative effect on the perturbation,so with suitable assumptions about scaling, an adjustment of −0.2×PS_(e)to the reference path would substantially cancel the perturbation seenby the ADC.

Alternative Embodiments

The person skilled in the art will realize that various rearrangementsof the diagrams provided herewith can be made without affecting theessential operation. For example, linear filtering operations can beinterchanged in order without affecting the transfer function; they canbe moved past addition or subtraction nodes provided suitablecompensating adjustments are made to the other paths. Two conceptuallydistinct filters may be combined into one, and in some cases, the filtermay disappear if the combined transfer function evaluates to unity.Addition and subtraction are essentially equivalent, if sign is adjustedelsewhere.

As an example, and without prejudice to the generality of the foregoing,in FIG. 3 the filter LPF⁻¹ may be deleted and an inverse filter LPFinserted following the simulator S. The balance between the simulatorpath and the measurement path is not disturbed by this change, and thesmall signal transfer function of the feedback loop may be restored toits previous value by a suitable adjustment to H.

Scaling factors have been omitted from this description for clarity.Signals, whether analog or digital, may be scaled according toconvenience: the multiplications, divisions, amplifications andattenuations that may be used to optimize scaling in a practicalimplementation have not been shown.

The use of a pair of on/off switches to represent a power switch shouldnot be taken as limiting. The skilled person will be able to adapt tothe principles described herein to full-bridge designs in which analogsignals are balanced, and to other more complex types of modulation. Thepower switch may similarly be replaced by a composite assembly thatitself includes feedback.

A number of exemplary embodiments having alternative structures will bedescribed below to illustrate some of the variations that are possiblewithin the scope of the invention.

One embodiment comprises an amplifier having: a pulse width modulatorreceiving a digital input; a power switch that produces a power switchoutput; and an output that is connected to the power switch output. Theamplifier also includes: a simulator that receives an input signalsubstantially similar to the signal received by the pulse widthmodulator; an ADC (analogue-to-digital converter) having an inputconnected to the power switch output; a subtractor having a first inputthat is connected to the output of the simulator, and a second inputthat is connected to the output of the ADC, where the input to the pulsewidth modulator is modified in dependence on the output of thesubtractor.

One embodiment also includes a noise shaper in which the simulator has afirst input and a second input, the second input being fed in dependenceon the output of the noise shaper. In one embodiment, the response ofthe simulator to its second input is substantially linear.

One embodiment also includes a low level switch whose switch timings arecontrolled by a digital modulator, wherein the input to the ADC isformed in dependence on the difference between the power switch outputand the output of the low level switch.

One embodiment also includes a low level switch that receives an inputfrom a second modulator. One embodiment also includes a second simulatorthat receives an input signal substantially similar to the signalreceived by the second modulator.

In one embodiment, the simulator models nonlinear characteristics of thepulse width modulator. In one embodiment, the simulator models knownimperfections of the power switch. In one embodiment, the simulatormodels delay or other allpass characteristics of the signal path fromthe input of the pulse width modulator to the second input of thesubtractor. In one embodiment, the simulator comprises an FIR filterwhose input is connected to a nonlinear function generator thatgenerates an arithmetic power of an input signal. In one embodiment, thesignal path from the power switch output to the ADC comprises a lowpassfilter.

In one embodiment, the signal path from the ADC to the second input ofthe subtractor comprises an FIR (finite impulse response) filter. In oneembodiment, the transfer function zeroes of the FIR filter substantiallycancel poles in the transfer function of the signal path from the powerswitch output to the input of the ADC.

One embodiment also includes a calibration unit that characterizes, atstart-up, the signal path from the power switch output to the input ofthe ADC, and configures at least one of: (i) the simulator; or (ii) thesignal path from the ADC to the second input of the subtractor; independence on the characterization. One embodiment includes acalibration unit that continuously characterizes the signal path fromthe power switch output to the input of the ADC, and adapts a parametergoverning the behaviour of: (i) the simulator; or (ii) the signal path;from the ADC to the second input of the subtractor responsively to thecharacterization. In one embodiment, the calibration unit adjustsparameters governing the amplifier behavior, where the adjustment ismade in response to a correlation calculated in dependence on the outputof the subtractor.

One embodiment also includes a filter in the signal path from thesubtractor to the input of the pulse width modulator, wherein the filteris substantially minimum phase and has an amplitude response that risesat frequencies above an operating frequency range.

One embodiment also includes a nonlinear correction unit in the signalpath from the subtractor to the input of the pulse width modulator, thecorrection unit substantially correcting, over an operating frequencyrange, a proportion of the nonlinear effects in the pulse widthmodulator. In one embodiment, the small signal transfer function of thenonlinear correction unit is minimum phase. One embodiment also includesa predistortion unit that substantially compensates the nonlineareffects in the pulse width modulator that are not compensated by thenonlinear correction unit. In one embodiment, the predistortion unit ismodified by at least low frequency components of a signal determined independence on the output of the subtractor.

One embodiment comprises an amplifier having: a digital modulator thatoperates at a first sampling frequency; power devices that produce anoutput; an ADC that receives an input in dependence on the output of thepower devices and that operates at a second sampling frequency that is amultiple of the first sampling frequency; and a decimator that receivesthe output of the ADC, the decimator comprising a decimation filter. Inthis embodiment, the input to the digital modulator is modified independence on the output of the of the decimator, and wherein thedecimation filter has transfer function zeroes some of which lie, in thecomplex z-plane, at positions significantly away from positions on theunit circle that correspond to the first sampling frequency and itsharmonics.

One embodiment comprises a switching amplifier having: a pulse widthmodulator that provides a low-level PWM waveform; a power switch thatproduces a power switch output; and an oversampling ADC followed by adecimator. In this embodiment, the ADC is responsive to the differencebetween a signal derived from the low-level PWM waveform and a signalderived from power switch output, and wherein the input of the powerswitch is responsive to the output of the decimator.

One embodiment comprises a switching amplifier having: a pulse widthmodulator that provides a low-level PWM waveform; a power switch thatproduces a power switch output; an ADC that is responsive to thedifference between a signal derived from the low-level PWM waveform anda signal derived from power switch output; and a shaping filter whoseinput is connected to the output of the ADC and whose response risesabove the operating frequency range. In this embodiment, the input ofthe power switch is responsive to the output of the shaping filter.

One embodiment comprises a switching amplifier having: a pulse widthmodulator that provides a low-level PWM waveform; a power switch thatproduces a power switch output; an ADC that is responsive to thedifference between a signal derived from the low-level PWM waveform anda signal derived from power switch output; and a low-delay correctorwhose input is responsive to the output of the ADC, the correctorproviding approximate or substantial correction, over the operatingfrequency range, for the nonlinear behavior of a pulse width modulator.In this embodiment, the input of the power switch is responsive to theoutput of the low-delay corrector.

Those of skill will appreciate that the various illustrative logicalblocks, modules, circuits, and algorithm steps described in connectionwith the embodiments disclosed herein may be implemented as electronichardware, computer software, or combinations of both. To clearlyillustrate this interchangeability of hardware and software, variousillustrative components, blocks, modules, circuits, and steps have beendescribed above generally in terms of their functionality. Whether suchfunctionality is implemented as hardware or software depends upon theparticular application and design constraints imposed on the overallsystem. Those of skill in the art may implement the describedfunctionality in varying ways for each particular application, but suchimplementation decisions should not be interpreted as causing adeparture from the scope of the present invention.

The various illustrative logical blocks, modules, and circuits describedin connection with the embodiments disclosed herein may be implementedor performed with application specific integrated circuits (ASICs),field programmable gate arrays (FPGAs), general purpose processors,digital signal processors (DSPs) or other logic devices, discrete gatesor transistor logic, discrete hardware components, or any combinationthereof designed to perform the functions described herein.

The steps of a method or algorithm described in connection with theembodiments disclosed herein may be embodied directly in hardware, insoftware (program instructions) executed by a processor, or in acombination of the two. Software may reside in RAM memory, flash memory,ROM memory, EPROM memory, EEPROM memory, registers, hard disk, aremovable disk, a CD-ROM, or any other form of storage medium known inthe art. Such a storage medium containing program instructions thatembody one of the present methods is itself an alternative embodiment ofthe invention. One exemplary storage medium may be coupled to aprocessor, such that the processor can read information from, and writeinformation to, the storage medium.

The benefits and advantages which may be provided by the presentinvention have been described above with regard to specific embodiments.These benefits and advantages, and any elements or limitations that maycause them to occur or to become more pronounced are not to be construedas critical, required, or essential features of any or all of theclaims. As used herein, the terms “comprises,” “comprising,” or anyother variations thereof, are intended to be interpreted asnon-exclusively including the elements or limitations which follow thoseterms. Accordingly, a system, method, or other embodiment that comprisesa set of elements is not limited to only those elements, and may includeother elements not expressly listed or inherent to the claimedembodiment.

While the present invention has been described with reference toparticular embodiments, it should be understood that the embodiments areillustrative and that the scope of the invention is not limited to theseembodiments. Many variations, modifications, additions and improvementsto the embodiments described above are possible. It is contemplated thatthese variations, modifications, additions and improvements fall withinthe scope of the invention as detailed within the following claims.

REFERENCES

-   1. Gerzon, M. A., “Predistortion Techniques for Complex but    Predictable Transmission Systems”, J. Audio Eng. Soc., Volume 20,    pp. 475-482 (July 1972).-   2. Bode, H. W., “Network Analysis and Feedback Amplifier Design”,    Litton, 1945; reprinted Van Nostrand, New York, 1959; reprinted    Krieger, New York 1975 ISBN 0-88275-242-1.-   3. Harris, S., Andersen, J., and Chieng, D., “Intelligent Class D    Amplifier Controller Integrated Circuit as an Ingredient Technology    for Multi-Channel Amplifier Modules of Greater than 50    Watts/Channel” Presented at the AES 115th Convention 2003 Oct. 10-13    New York, Audio Eng. Soc. preprint #5947-   4. Craven, P. G., “Toward the 24-bit DAC: Novel Noise-Shaping    Topologies Incorporating Correction for the Nonlinearity in a PWM    Output Stage”, J. Audio Eng. Soc., Volume 41 Number 5 pp. 291-313;    May 1993-   5. Hawksford, M. O. J, “Dynamic Model-Based Linearization of    Quantized Pulse-Width Modulation for Applications in    Digital-to-Analog Conversion and Digital Power Amplifier    Systems”, J. Audio Eng. Soc., Volume 40 Number 4 pp. 235-252; April    1992-   6. Sandler, M., “Towards a Digital Power Amplifier” Audio Eng. Soc    Preprint Number: 2135, September 1984-   7. Norsworthy, S. R., Schreier, R, Temes, G. C. (editors),    “Delta-Sigma Data Converters: Theory, Design and Simulation”, IEEE    Press 1997, ISBN 0-7803-1045-4.

1. A low delay corrector (LDC) unit, comprising: a nonlinear functiongenerator that receives a first signal and that outputs a second signalin dependence on the first signal and a transfer function of thenonlinear function generator; and a filter fed in dependence on thesecond signal output by the nonlinear function generator; wherein thefirst signal received by the nonlinear function generator is derived independence on an input signal provided to an input of the LDC unit andan output of the filter; and wherein an output of the LDC unit isderived in dependence on the first signal received by the nonlinearfunction generator and the second signal output by the nonlinearfunction generator.
 2. A low delay corrector (LDC) unit according toclaim 1, wherein: the input signal provided to the input of the LDC is adigital audio signal.
 3. A low delay corrector (LDC) unit according toclaim 1, wherein the LDC unit is configured to: be placed in a signalpath of a plant including a pulse width modulator; and compensate for atleast some nonlinear effects of the pulse width modulator over anoperating frequency range.
 4. A low delay corrector (LDC) unit accordingto claim 1, wherein: the filter comprises a finite impulse response(FIR) filter.
 5. A low delay corrector (LDC) unit according to claim 4,wherein: a z-transform of the FIR filter comprises (5*z⁻¹−4*z⁻²+z⁻³)/2.6. A low delay corrector (LDC) unit according to claim 1, wherein: thenonlinear function generator implements a cubic polynomial function. 7.A low delay corrector (LDC) unit according to claim 1, wherein: theoutput of the LDC is derived as the difference between the first signalreceived by the nonlinear function generator and the second signaloutput by the nonlinear function generator.
 8. A low delay corrector(LDC) unit according to claim 1, wherein the nonlinear functiongenerator is configured to output the second signal in dependence on thefirst signal using the transfer function: P(y)=y³/12, where y is thefirst signal received by the nonlinear function generator, and P(y) isthe second signal output by the non-linear function generator.
 9. A lowdelay corrector (LDC) unit according to claim 1, wherein the nonlinearfunction generator is configured to output the second signal independence on the first signal using the transfer function:P(y)=0.08251487120 y³−0.01495088616 y⁵, where y is the first signalreceived by the nonlinear function generator, and P(y) is the secondsignal output by the non-linear function generator.
 10. A low delaycorrector (LDC) unit according to claim 1, further comprising: a clipperhaving an input that receives the second signal output by the nonlinearfunction generator and an output that feeds the filter.
 11. A method forlow delay correction, comprising: (a) producing a second signal independence on a first signal and a nonlinear transfer function; (b)feeding a filter in dependence on the second signal; (c) deriving thefirst signal in dependence on an input signal and an output of thefilter; and (d) deriving an output signal in dependence on the firstsignal and the second signal.
 12. The method of claim 11, wherein: theinput signal comprises a digital audio signal; and step (c) comprisesderiving the first signal in dependence on the digital audio signal andthe output of the filter.
 13. The method of claim 11, wherein saidmethod is for use by a plant including a pulse width modulator, tocompensate for at least some nonlinear effects of the pulse widthmodulator over an operating frequency range.
 14. The method of claim 11,wherein: the filter comprises a finite impulse response (FIR) filter;step (b) comprises feeding the impulse response filter in dependence onthe second signal; and step (c) comprises deriving the first signal independence on the input signal and the output of the impulse responsefilter.
 15. The method of claim 14, wherein: a z-transform response ofthe FIR filter comprises (5*z⁻¹−4*z⁻²+z⁻³)/2.
 16. The method of claim11, wherein: the nonlinear function comprises a cubic polynomialfunction; and step (a) comprises producing the second signal independence on the first signal and the cubic polynomial function. 17.The method of claim 11, wherein: step (d) comprises deriving the outputsignal as the difference between the first signal and the second signal.18. The method of claim 11, wherein: step (a) comprises using thenonlinear transfer function P(y)=y³/12 to produce the second signal,where P(y) is the second signal and y is the first signal.
 19. A methodof claim 11, wherein: step (a) comprises using the nonlinear transferfunction P(y)=0.08251487120 y³−0.01495088616 y⁵ to produce the secondsignal, where P(y) is the second signal and y is the first signal. 20.The method of claim 11, further comprising: limiting the second signalto a range prior to feeding the filter in dependence on the secondsignal.
 21. The method of claim 20, wherein: the limiting compriseslimiting the second signal to levels of approximately +/−0.125 prior tofeeding the filter in dependence on the second signal.